Re: Phase Reversal on Santa Cruz (cs46xx)

Re: Phase Reversal on Santa Cruz (cs46xx)

[Kevin Puetz]

On Sunday 17 November 2002 00:56, you wrote:
> I am not currently a subscriber to this list, so pease CC me on any
> responses.

OK,  I decided to subscribe, so I ignore this.

> Frank, Benny, the guess of a phase reversal as responsible for the 'lack of
> bass' on this card is dead-on - a little tinkering with artsbuilder (had to
> remember how to do it) to place an inverter in one channel of the output
> path brings sound back to perfection. Great guess!
>
> I can pretty much leave the sub (which has one cone and mixes the channels
> first) silent against a full-volume sine wave, or invert it and have it
> perfect. I can definitely hear sine waves at higher frequencies attenuated
> to some extent if I move the speakers directly adjacent to one another. I'd
> say that confirms it pretty well.

I have also got an aRts stereo effect to fix it for noatun now, so I can 
verify it fixing the sound for general music as well. Yippie :-)

> So... anything that can be done to fix this (I assume so, since it's a new
> problem since older alsa), or do I have to cut my speaker wires apart? :-P

Still hopfully alsa gets fixed and I can throw away the effect plugin :-)


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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Benny Sjostrand]

>
>
>>So... anything that can be done to fix this (I assume so, since it's a new
>>problem since older alsa), or do I have to cut my speaker wires apart? :-P
>>    
>>
I'm still a newbie to the audio world, but just wondering, why design a 
audio equipment
that behaves like this ??
(When you run out of ideas howto fix something let's try to move it into 
a HW bug instead -;) )

>Still hopfully alsa gets fixed and I can throw away the effect plugin :-)
>
>
>  
>
One alternative solution would be program the DSP to do the job instead 
and you will
save CPU time. (Yet another thing that I dont known howto do -:(  )

/Benny

PS. being out for traveling some days (wont respond any mail for a while 
...)



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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Kevin Puetz]

On Sunday 17 November 2002 23:01, Benny Sjostrand wrote:
> >>So... anything that can be done to fix this (I assume so, since it's a
> >> new problem since older alsa), or do I have to cut my speaker wires
> >> apart? :-P
>
> I'm still a newbie to the audio world, but just wondering, why design a
> audio equipment
> that behaves like this ??
> (When you run out of ideas howto fix something let's try to move it into
> a HW bug instead -;) )

well, the basic premise is simple... the speaker wires have a positive and 
negative wire, current flows through the coil, and the cone moves one way or 
the other. If you swap the wires, the cone moves the other way. Inaudible, 
except that if the two speakers aren't moving the same way, they waves tend 
to cancel each other out. It affects bass more than treble because the waves 
are longer and slower, so a) they correleation between the stereo channels is 
stronger and b) inaccuracies in soundcard timing, etc have less impact. The 
fact that sound is sensitive to polarity is just basic physics :-)

Now, why in the world the santa cruz would be wired in such a way that 
software which is fine on other cards would get one channel backwards on it 
is an excellent question. Espescially when there is older software that 
worked on all (though not so well, I definitely like having multiple apps 
accessing it at once).

> >Still hopfully alsa gets fixed and I can throw away the effect plugin :-)
>
> One alternative solution would be program the DSP to do the job instead
> and you will
> save CPU time. (Yet another thing that I dont known howto do -:(  )

well, how did it work before (0.9.0rc1 worked for sure, I guess I should try 
to identify the specific version that developed this problem)? Were we using 
the DSP to invert it previosly, did we lose some Santa Cruz specific 
bittwiddling (the 'fix' consists of just negating all the levels for one 
stereo channel). Any guesses?

> /Benny
>
> PS. being out for traveling some days (wont respond any mail for a while
> ...)

Just as another question, do anyone know if it's possible to use the internal 
SPDIF connector to feed CD audio into the card instead of the analog input? I 
can't find anything in the mixer that seems to recieve that input.


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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Michael Alan Dorman]

Kevin Puetz <puetzk@iastate.edu> writes:
> Now, why in the world the santa cruz would be wired in such a way that 
> software which is fine on other cards would get one channel backwards on it 
> is an excellent question. Espescially when there is older software that 
> worked on all (though not so well, I definitely like having multiple apps 
> accessing it at once).

It's not just the Santa Cruz, though---my IBM T22 laptop has the exact
same problem.

Mike.


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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Kevin Puetz]

On Monday 18 November 2002 08:30, Michael Alan Dorman wrote:
> Kevin Puetz <puetzk@iastate.edu> writes:
> > Now, why in the world the santa cruz would be wired in such a way that
> > software which is fine on other cards would get one channel backwards on
> > it is an excellent question. Espescially when there is older software
> > that worked on all (though not so well, I definitely like having multiple
> > apps accessing it at once).
>
> It's not just the Santa Cruz, though---my IBM T22 laptop has the exact
> same problem.

bizzarre...

okay, well for any aRts/noatun users who'd like to a) test their card or b) 
get a temporary workaround, I'll upload my cheesy effect fikter, which will 
either make your sound much better or much worse :-).

http://ophelia.student.iastate.edu/~puetzk/arts_phaseinvert-0.5.tgz

standard autoconf/automake setup for kde3, no docs :-)
After installing it into your KDE prefix noatun should show 
Synth_STEREO_PHASEINVERSION as a choice in it's effects config dialog.

> Mike.
>
>
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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Kevin Puetz]

On Monday 18 November 2002 22:00, Weijia Yang wrote:
> > okay, well for any aRts/noatun users who'd like to a) test their card or
> > b) get a temporary workaround, I'll upload my cheesy effect fikter, which
> > will either make your sound much better or much worse :-).
> >
> > http://ophelia.student.iastate.edu/~puetzk/arts_phaseinvert-0.5.tgz
> >
> > standard autoconf/automake setup for kde3, no docs :-)
> > After installing it into your KDE prefix noatun should show
> > Synth_STEREO_PHASEINVERSION as a choice in it's effects config dialog.
> >
> > > Mike.
>
> I am not quite sure how to go about compiling this effects plugin.  Could
> you give me a short rundown? Thanks
>
> Weijia Yang

Alright, standard kde autoconf directions as follows :-P

you will need a recent version of autotools (automake, autoconf, autoheader, 
etc), arts and kdelibs development headers from kde 3 (or 3.1) (ie, you need 
the -dev pacakges). That should be all... then

make -f Makefile.cvs
./configure --prefix=`kde-config --prefix`
make
make install

that should be all, it built here... what problem are you having specifically? 


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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Kevin Puetz]

On Tuesday 19 November 2002 01:28, you wrote:
> I get the following error during the 'make' step.  I am running kde3.1 rc3
> with gcc3.2

err, oops. It only built here because my standard artsmodules.h had stuff in 
it from when I'd originally just added it to kdemm. My bad.

grab http://ophelia.student.iastate.edu/~puetzk/arts_phaseinvert-0.6.tgz, I 
fixed it.


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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Benny Sjostrand]

>
>
>well, the basic premise is simple... the speaker wires have a positive and 
>negative wire, current flows through the coil, and the cone moves one way or 
>the other. If you swap the wires, the cone moves the other way. Inaudible, 
>except that if the two speakers aren't moving the same way, they waves tend 
>to cancel each other out. It affects bass more than treble because the waves 
>are longer and slower, so a) they correleation between the stereo channels is 
>stronger and b) inaccuracies in soundcard timing, etc have less impact. The 
>fact that sound is sensitive to polarity is just basic physics :-)
>  
>
Thanks, that explication makes sense.

>Now, why in the world the santa cruz would be wired in such a way that 
>software which is fine on other cards would get one channel backwards on it 
>is an excellent question. Espescially when there is older software that 
>worked on all (though not so well, I definitely like having multiple apps 
>accessing it at once).
>  
>
If it was that simple that Santa Cruz where the only card in the world with
this problem, but there's the same problem is reported for other cs46xx 
based card's
too.

>well, how did it work before (0.9.0rc1 worked for sure, I guess I should try 
>to identify the specific version that developed this problem)? Were we using 
>the DSP to invert it previosly, did we lose some Santa Cruz specific 
>bittwiddling (the 'fix' consists of just negating all the levels for one 
>stereo channel). Any guesses?
>  
>
I would like know why the problem does not apear with rc1,  but the 
problem does not
apear  forme with any release, so I cant reproduce the problem with my 
equipment, which
dont make things easy ...

The main diference between the old DSP code and the new is that the old DSP
is based on a static parameter setup. The new DSP code is  a attempt to 
manage
tasks and resources in the DSP dynamically.

I dont find any reason to why the DSP would invert one of the channels, 
in theory
the parameter setup made when playing one PCM channel is the same as it is
in the old DSP code static loaded (discarding unknown bugs)

Finally, the "volume" could be possible, speculating: if the output sample
is the product of volume * sample, and the volume is negative the sample 
would
be negated. The "volume" in a DSP SCB for each channel is 16 bit value where
0x8000 is the maximum volume and 0xffff is cero, and lower then 0x8000 dont
know. This part now actually differs a little bit from the old static 
DSP setup.
Have you tested my latest patch ??

>Just as another question, do anyone know if it's possible to use the internal 
>SPDIF connector to feed CD audio into the card instead of the analog input? I 
>can't find anything in the mixer that seems to recieve that input.
>  
>

The cs4630 chip got only one SDPIF input interface, you should see it 
the mixer with
the new DSP code. And if that interface is wired to the internal SPDIF 
conector without
any GPIO story it should just work (however the sounds gets 
distorcionated (out-of-sync)
sometimes with the SPDIF input) ...

/Benny



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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Kevin Puetz]

On Thursday 21 November 2002 18:09, Benny Sjostrand wrote:
> Thanks, that explication makes sense.
>
> If it was that simple that Santa Cruz where the only card in the world with
> this problem, but there's the same problem is reported for other cs46xx
> based card's
> too.
>
> >well, how did it work before (0.9.0rc1 worked for sure, I guess I should
> > try to identify the specific version that developed this problem)? Were
> > we using the DSP to invert it previosly, did we lose some Santa Cruz
> > specific bittwiddling (the 'fix' consists of just negating all the levels
> > for one stereo channel). Any guesses?
>
> I would like know why the problem does not apear with rc1,  but the
> problem does not
> apear  forme with any release, so I cant reproduce the problem with my
> equipment, which
> dont make things easy ...

no, that would be annoying indeed :-)

> The main diference between the old DSP code and the new is that the old DSP
> is based on a static parameter setup. The new DSP code is  a attempt to
> manage
> tasks and resources in the DSP dynamically.
>
> I dont find any reason to why the DSP would invert one of the channels,
> in theory
> the parameter setup made when playing one PCM channel is the same as it is
> in the old DSP code static loaded (discarding unknown bugs)
>
> Finally, the "volume" could be possible, speculating: if the output sample
> is the product of volume * sample, and the volume is negative the sample
> would
> be negated. The "volume" in a DSP SCB for each channel is 16 bit value
> where 0x8000 is the maximum volume and 0xffff is cero, and lower then
> 0x8000 dont know. This part now actually differs a little bit from the old
> static DSP setup.
> Have you tested my latest patch ??

no, I haven't, since the patch is quite mangled in the www archive and I 
wasn't subscribed then, I didn't go through th trouble of fixing it up. Is it 
merged into cvs now, and if not, could you send it my way? I'd be happy to 
give it a shot.

> >Just as another question, do anyone know if it's possible to use the
> > internal SPDIF connector to feed CD audio into the card instead of the
> > analog input? I can't find anything in the mixer that seems to recieve
> > that input.
>
> The cs4630 chip got only one SDPIF input interface, you should see it
> the mixer with
> the new DSP code. And if that interface is wired to the internal SPDIF
> conector without
> any GPIO story it should just work (however the sounds gets
> distorcionated (out-of-sync)
> sometimes with the SPDIF input) ...

Hmm, OK. Guess I'll reboot to windows and see if that connection even works 
there :-P Maybe I just screwed it up.

> /Benny

oh, for anybody using this
 http://ophelia.student.iastate.edu/~puetzk/arts_phaseinvert-0.7.tgz

fixing yet another silly glitch Weijia found. (This is an aRts effect filter 
which introduces a phase inversion between the channels on purpose, which 
either fixes or introduces the problem, depending on whether you had it 
already).


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Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[Benny Sjostrand]

[-- Attachment #1: Type: text/plain, Size: 318 bytes --]

>
>
>no, I haven't, since the patch is quite mangled in the www archive and I 
>wasn't subscribed then, I didn't go through th trouble of fixing it up. Is it 
>merged into cvs now, and if not, could you send it my way? I'd be happy to 
>give it a shot.
>  
>

Here is the patch again, it's not in current CVS.

/Benny

[-- Attachment #2: cs46xx.patch --]
[-- Type: text/plain, Size: 19101 bytes --]

diff --exclude=CVS -Naur alsa-kernel/include/cs46xx_dsp_spos.h ../cvs/alsa-kernel/include/cs46xx_dsp_spos.h
--- alsa-kernel/include/cs46xx_dsp_spos.h	Sun Nov 10 19:47:01 2002
+++ ../cvs/alsa-kernel/include/cs46xx_dsp_spos.h	Thu Nov 14 00:39:06 2002
@@ -63,7 +63,6 @@
 #define DSP_SPDIF_STATUS_OUTPUT_ENABLED 1
 #define DSP_SPDIF_STATUS_PLAYBACK_OPEN  2
 #define DSP_SPDIF_STATUS_HW_ENABLED     4
-#define DSP_SPDIF_STATUS_AC3_MODE       8
 
 struct _dsp_module_desc_t;
 
@@ -196,6 +195,10 @@
 	int spdif_status_in;
 	u16 spdif_input_volume_right;
 	u16 spdif_input_volume_left;
+	/* spdif channel status,
+	   left right and user validity bits */
+	int spdif_csuv_default;
+	int spdif_csuv_stream;
 
 	/* SPDIF input sample rate converter */
 	dsp_scb_descriptor_t * spdif_in_src;
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/cs46xx_lib.c ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.c
--- alsa-kernel/pci/cs46xx/cs46xx_lib.c	Mon Nov 11 23:04:50 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.c	Sun Nov 17 16:26:53 2002
@@ -946,15 +946,15 @@
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
+		/* magic value to unmute PCM stream  playback volume */
+		snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 
+				       SCBVolumeCtrl) << 2, 0x80007fff);
+
 		if (cpcm->pcm_channel->unlinked)
 			cs46xx_dsp_pcm_link(chip,cpcm->pcm_channel);
+
 		if (substream->runtime->periods != CS46XX_FRAGS)
 			snd_cs46xx_playback_transfer(substream, 0);
-
-		/* raise playback volume */
-		cs46xx_dsp_scb_set_volume (chip,cpcm->pcm_channel->pcm_reader_scb,
-					   chip->dsp_spos_instance->dac_volume_right,
-					   chip->dsp_spos_instance->dac_volume_left);
 #else
 		if (substream->runtime->periods != CS46XX_FRAGS)
 			snd_cs46xx_playback_transfer(substream, 0);
@@ -968,8 +968,9 @@
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-		/* mute channel */
-		cs46xx_dsp_scb_set_volume (chip,cpcm->pcm_channel->pcm_reader_scb,0,0);
+		/* magic mute channel */
+		snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 
+				       SCBVolumeCtrl) << 2, 0xffffffff);
 
 		if (!cpcm->pcm_channel->unlinked)
 			cs46xx_dsp_pcm_unlink(chip,cpcm->pcm_channel);
@@ -1063,6 +1064,7 @@
 }
 #endif
 
+
 static int snd_cs46xx_playback_hw_params(snd_pcm_substream_t * substream,
 					 snd_pcm_hw_params_t * hw_params)
 {
@@ -1079,15 +1081,10 @@
 	down (&chip->spos_mutex);
 
 	snd_assert (cpcm->pcm_channel != NULL);
-
-	/* if IEC958 is opened in AC3 mode dont adjust SRCTask is not
-	   used so dont adjust sample rate */
-	if (cpcm->pcm_channel->pcm_channel_id != DSP_IEC958_CHANNEL ||
-	    !(chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE)) {
-		if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
-			up (&chip->spos_mutex);
-			return -ENXIO;
-		}
+	
+	if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
+		up (&chip->spos_mutex);
+		return -ENXIO;
 	}
 
 	if (cs46xx_dsp_pcm_channel_set_period (chip,cpcm->pcm_channel,period_size * 4)) {
@@ -1886,8 +1883,8 @@
 {
 	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
 
-	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_right;
-	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_left;
+	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_left;
+	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_right;
 
 	return 0;
 }
@@ -2038,34 +2035,6 @@
 	return 0;
 }
 
-static int snd_cs46xx_iec958_ac3_mode_get(snd_kcontrol_t *kcontrol, 
-					  snd_ctl_elem_value_t *ucontrol) 
-{
-	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
-	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
-
-	if (!ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) 
-		ucontrol->value.integer.value[0] = 1;
-	else
-		ucontrol->value.integer.value[0] = 0;
-
-	return 0;
-}
-
-static int snd_cs46xx_iec958_ac3_mode_put(snd_kcontrol_t *kcontrol, 
-                                      snd_ctl_elem_value_t *ucontrol) 
-{
-	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
-	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
-	int old = ins->spdif_status_out;
-
-	if (ucontrol->value.integer.value[0]) 
-		ins->spdif_status_out |= DSP_SPDIF_STATUS_AC3_MODE;
-	else
-		ins->spdif_status_out &= ~DSP_SPDIF_STATUS_AC3_MODE;
-
-	return (old != ins->spdif_status_out);
-}
 
 static int snd_cs46xx_pcm_capture_put(snd_kcontrol_t *kcontrol, 
                                       snd_ctl_elem_value_t *ucontrol)
@@ -2127,6 +2096,110 @@
 	return (val1 != snd_cs46xx_peekBA0(chip, BA0_EGPIODR));
 }
 
+
+static int snd_cs46xx_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+	return 0;
+}
+
+static int snd_cs46xx_spdif_default_get(snd_kcontrol_t * kcontrol,
+					snd_ctl_elem_value_t * ucontrol)
+{
+	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
+
+	down (&chip->spos_mutex);
+	ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_default >> 24) & 0xff);
+	ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_default >> 16) & 0xff);
+	ucontrol->value.iec958.status[2] = 0;
+	ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_default) & 0xff);
+	up (&chip->spos_mutex);
+
+	return 0;
+}
+
+static int snd_cs46xx_spdif_default_put(snd_kcontrol_t * kcontrol,
+					snd_ctl_elem_value_t * ucontrol)
+{
+	cs46xx_t * chip = snd_kcontrol_chip(kcontrol);
+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
+	unsigned int val;
+	int change;
+
+	down (&chip->spos_mutex);
+	val = _wrap_all_bits(((u32)ucontrol->value.iec958.status[0] << 24)) |
+	      _wrap_all_bits(((u32)ucontrol->value.iec958.status[2] << 16)) |
+	      _wrap_all_bits( (u32)ucontrol->value.iec958.status[3])  |
+		/* left and right validity bit */
+		(1 << 13) | (1 << 12);
+
+
+	change = ins->spdif_csuv_default != val;
+	ins->spdif_csuv_default = val;
+
+	if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) )
+		cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val);
+
+	up (&chip->spos_mutex);
+
+	return change;
+}
+
+static int snd_cs46xx_spdif_mask_get(snd_kcontrol_t * kcontrol,
+				     snd_ctl_elem_value_t * ucontrol)
+{
+	ucontrol->value.iec958.status[0] = 0xff;
+	ucontrol->value.iec958.status[1] = 0xff;
+	ucontrol->value.iec958.status[2] = 0x00;
+	ucontrol->value.iec958.status[3] = 0xff;
+	return 0;
+}
+
+static int snd_cs46xx_spdif_stream_get(snd_kcontrol_t * kcontrol,
+                                         snd_ctl_elem_value_t * ucontrol)
+{
+	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
+
+	down (&chip->spos_mutex);
+	ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_stream >> 24) & 0xff);
+	ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_stream >> 16) & 0xff);
+	ucontrol->value.iec958.status[2] = 0;
+	ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_stream) & 0xff);
+	up (&chip->spos_mutex);
+
+	return 0;
+}
+
+static int snd_cs46xx_spdif_stream_put(snd_kcontrol_t * kcontrol,
+                                        snd_ctl_elem_value_t * ucontrol)
+{
+	cs46xx_t * chip = snd_kcontrol_chip(kcontrol);
+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
+	unsigned int val;
+	int change;
+
+	down (&chip->spos_mutex);
+	val = _wrap_all_bits(((u32)ucontrol->value.iec958.status[0] << 24)) |
+	      _wrap_all_bits(((u32)ucontrol->value.iec958.status[1] << 16)) |
+	      _wrap_all_bits( (u32)ucontrol->value.iec958.status[3]) |
+		/* left and right validity bit */
+		(1 << 13) | (1 << 12);
+
+
+	change = ins->spdif_csuv_stream != val;
+	ins->spdif_csuv_stream = val;
+
+	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN )
+		cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val);
+
+	up (&chip->spos_mutex);
+
+	return change;
+}
+
 #endif /* CONFIG_SND_CS46XX_NEW_DSP */
 
 #ifdef CONFIG_SND_CS46XX_DEBUG_GPIO
@@ -2238,7 +2311,7 @@
 },
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name = "IEC 958 Output Switch",
+	.name = "IEC958 Output Switch",
 	.info = snd_mixer_boolean_info,
 	.get = snd_cs46xx_iec958_get,
 	.put = snd_cs46xx_iec958_put,
@@ -2246,14 +2319,7 @@
 },
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name = "IEC 958 AC3 Mode Switch",
-	.info = snd_mixer_boolean_info,
-	.get = snd_cs46xx_iec958_ac3_mode_get,
-	.put = snd_cs46xx_iec958_ac3_mode_put,
-},
-{
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name = "IEC 958 Input Switch",
+	.name = "IEC958 Input Switch",
 	.info = snd_mixer_boolean_info,
 	.get = snd_cs46xx_iec958_get,
 	.put = snd_cs46xx_iec958_put,
@@ -2261,12 +2327,34 @@
 },
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name = "IEC 958 Input Volume",
+	.name = "IEC958 Input Volume",
 	.info = snd_cs46xx_vol_info,
 	.get = snd_cs46xx_vol_iec958_get,
 	.put = snd_cs46xx_vol_iec958_put,
 	.private_value = (ASYNCRX_SCB_ADDR + 0xE) << 2,
 },
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.name =  SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+	.info =	 snd_cs46xx_spdif_info,
+	.get =	 snd_cs46xx_spdif_default_get,
+	.put =   snd_cs46xx_spdif_default_put,
+},
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.name =	 SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+	.info =	 snd_cs46xx_spdif_info,
+        .get =	 snd_cs46xx_spdif_mask_get,
+	.access = SNDRV_CTL_ELEM_ACCESS_READ
+},
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+	.name =	 SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+	.info =	 snd_cs46xx_spdif_info,
+	.get =	 snd_cs46xx_spdif_stream_get,
+	.put =	 snd_cs46xx_spdif_stream_put
+},
+
 #endif
 #ifdef CONFIG_SND_CS46XX_DEBUG_GPIO
 {
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/cs46xx_lib.h ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.h
--- alsa-kernel/pci/cs46xx/cs46xx_lib.h	Sun Nov 10 19:47:01 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.h	Wed Nov 13 22:51:21 2002
@@ -217,6 +217,6 @@
 							       int period_size);
 int                        cs46xx_dsp_pcm_ostream_set_period (cs46xx_t * chip,
 							      int period_size);
-int                        cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 right,u16 left);
-int                        cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 right,u16 left);
+int                        cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 left,u16 right);
+int                        cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 left,u16 right);
 #endif /* __CS46XX_LIB_H__ */
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos.c ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.c
--- alsa-kernel/pci/cs46xx/dsp_spos.c	Sun Nov 10 19:47:01 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.c	Thu Nov 14 19:59:59 2002
@@ -31,6 +31,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/info.h>
+#include <sound/asoundef.h>
 #include <sound/cs46xx.h>
 
 #include "cs46xx_lib.h"
@@ -262,6 +263,15 @@
 	ins->spdif_input_volume_right = 0x8000;
 	ins->spdif_input_volume_left = 0x8000;
 
+	/* set left and right validity bits and
+	   default channel status */
+	ins->spdif_csuv_default = 
+		ins->spdif_csuv_stream =  
+	 /* byte 0 */  (_wrap_all_bits(  (SNDRV_PCM_DEFAULT_CON_SPDIF        & 0xff)) << 24) |
+	 /* byte 1 */  (_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 16) & 0xff)) << 16) |
+	 /* byte 3 */   _wrap_all_bits(  (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) |
+	 /* left and right validity bits */ (1 << 13) | (1 << 12);
+
 	return ins;
 }
 
@@ -1549,7 +1559,7 @@
 	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CONTROL, 0x80000000);
 
 	/* right and left validate bit */
-	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x00000000 | (1 << 13) | (1 << 12));
+	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
 
 	/* monitor state */
 	ins->spdif_status_out |= DSP_SPDIF_STATUS_HW_ENABLED;
@@ -1587,11 +1597,6 @@
 	cs46xx_poke_via_dsp (chip,SP_SPDIN_FIFOPTR, 0x0);
 	cs46xx_src_link(chip,ins->spdif_in_src);
 
-	/* restore SPDIF input volume */
-	cs46xx_dsp_scb_set_volume (chip,ins->spdif_in_src,
-				   ins->spdif_input_volume_right,
-				   ins->spdif_input_volume_left);
-
 	spin_unlock_irq(&chip->reg_lock);
 
 	/* set SPDIF input sample rate and unmute
@@ -1725,39 +1730,47 @@
 	return 0;
 }
 
-int cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 right,u16 left)
+int cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 left,u16 right)
 {
-	int i;
 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
+	dsp_scb_descriptor_t * scb; 
 
 	down(&chip->spos_mutex);
+	
+	/* main output */
+	scb = ins->master_mix_scb->sub_list_ptr;
+	while (scb != ins->the_null_scb) {
+		cs46xx_dsp_scb_set_volume (chip,scb,left,right);
+		scb = scb->next_scb_ptr;
+	}
 
-	ins->dac_volume_right = right;
-	ins->dac_volume_left = left;
-
-	for (i = 0; i < DSP_MAX_PCM_CHANNELS; ++i) {
-		if (ins->pcm_channels[i].active &&
-		    !ins->pcm_channels[i].unlinked) {
-			cs46xx_dsp_scb_set_volume (chip,ins->pcm_channels[i].pcm_reader_scb,
-						   right,left);
-			
-		}
+	/* rear output */
+	scb = ins->rear_mix_scb->sub_list_ptr;
+	while (scb != ins->the_null_scb) {
+		cs46xx_dsp_scb_set_volume (chip,scb,left,right);
+		scb = scb->next_scb_ptr;
 	}
 
+	ins->dac_volume_left = left;
+	ins->dac_volume_right = right;
+
 	up(&chip->spos_mutex);
 
 	return 0;
 }
 
-int cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 right,u16 left) {
+int cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 left,u16 right) {
 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
 
 	down(&chip->spos_mutex);
-	cs46xx_dsp_scb_set_volume (chip,ins->spdif_in_src,
-				   right,left);
 
-	ins->spdif_input_volume_right = right;
+	if (ins->asynch_rx_scb != NULL)
+		cs46xx_dsp_scb_set_volume (chip,ins->asynch_rx_scb,
+					   left,right);
+
 	ins->spdif_input_volume_left = left;
+	ins->spdif_input_volume_right = right;
+
 	up(&chip->spos_mutex);
 
 	return 0;
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos.h ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.h
--- alsa-kernel/pci/cs46xx/dsp_spos.h	Sun Nov 10 19:47:01 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.h	Thu Nov 14 09:36:26 2002
@@ -185,6 +185,25 @@
 #define SP_SPDOUT_CONTROL 0x804D
 #define SP_SPDOUT_CSUV    0x808E
 
+static inline u8 _wrap_all_bits (u8 val) {
+	u8 wrapped;
+	
+	/* wrap all 8 bits */
+	wrapped = 
+		((val & 0x1 ) << 7) |
+		((val & 0x2 ) << 5) |
+		((val & 0x4 ) << 3) |
+		((val & 0x8 ) << 1) |
+		((val & 0x10) >> 1) |
+		((val & 0x20) >> 3) |
+		((val & 0x40) >> 5) |
+		((val & 0x80) >> 6);
+
+	return wrapped;
+
+}
+
+
 static inline void cs46xx_dsp_spos_update_scb (cs46xx_t * chip,dsp_scb_descriptor_t * scb) 
 {
 	/* update nextSCB and subListPtr in SCB */
@@ -195,12 +214,11 @@
 }
 
 static inline void cs46xx_dsp_scb_set_volume (cs46xx_t * chip,dsp_scb_descriptor_t * scb,
-					      u16 right,u16 left) {
-	unsigned int val = ((0xffff - right) << 16 | (0xffff - left));	
+					      u16 left,u16 right) {
+	unsigned int val = ((0xffff - left) << 16 | (0xffff - right));
 
 	snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val);
 	snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val);
 }
-
 #endif /* __DSP_SPOS_H__ */
 #endif /* CONFIG_SND_CS46XX_NEW_DSP  */
diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c ../cvs/alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c
--- alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c	Sun Nov 10 19:47:01 2002
+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c	Thu Nov 14 09:08:14 2002
@@ -603,8 +603,8 @@
 		src_buffer_addr << 0x10,
 		0x04000000,
 		{ 
-			0x8000,0x8000,
-			0xffff,0xffff
+			0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left,
+			0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left
 		}
 	};
 
@@ -658,7 +658,7 @@
 		/* D */ 0,
 		{
 			/* E */ 0x8000,0x8000,
-			/* F */ 0xffff,0xffff
+			/* F */ 0x8000,0x8000
 		}
 	};
 
@@ -830,7 +830,7 @@
 		0,0x2aab,           /* Const 1/3 */
     
 		{
-			0,                /* Define the unused elements */
+			0,         /* Define the unused elements */
 			0,
 			0
 		},
@@ -846,7 +846,7 @@
 		   rate etc  */
 		0x18000000,                     /* Phi increment for approx 32k operation */
 		0x8000,0x8000,                  /* Volume controls are unused at this time */
-		0xffff,0xffff
+		0x8000,0x8000
 	};
   
 	scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_tx_scb,
@@ -864,7 +864,7 @@
                                    dsp_scb_descriptor_t * parent_scb,
                                    int scb_child_type)
 {
-
+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
 	dsp_scb_descriptor_t * scb;
 
 	asynch_fg_rx_scb_t asynch_fg_rx_scb = {
@@ -893,9 +893,9 @@
 		   rate etc  */
 		0x18000000,         
 
-		/* Mute stream */
-		0x8000,0x8000,       
-		0xffff,0xffff
+		/* Set IEC958 input volume */
+		0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left,
+		0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left,
 	};
 
 	scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_rx_scb,
@@ -1116,11 +1116,13 @@
 	case DSP_IEC958_CHANNEL:
 		snd_assert (ins->asynch_tx_scb != NULL, return NULL);
 		mixer_scb = ins->asynch_tx_scb;
+#if 0
 		if (ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) {
 			snd_printdd ("IEC958 opened in AC3 mode\n");
 			/*src_scb = ins->asynch_tx_scb;
 			  ins->asynch_tx_scb->ref_count ++;*/
 		}
+#endif
 		break;
 	default:
 		snd_assert (0);
@@ -1198,9 +1200,7 @@
 			return NULL;
 		}
 
-		if (pcm_channel_id != DSP_IEC958_CHANNEL ||
-		    !(ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE))
-			cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate);
+		cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate);
 
 		ins->nsrc_scb ++;
 	} 
@@ -1461,17 +1461,11 @@
 	 */
 	spin_lock_irqsave(&chip->reg_lock, flags);
 
-	/* mute SCB */
-	/* cs46xx_dsp_scb_set_volume (chip,src,0,0); */
-
 	snd_cs46xx_poke(chip, (src->address + SRCCorPerGof) << 2,
 	  ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));
 
 	snd_cs46xx_poke(chip, (src->address + SRCPhiIncr6Int26Frac) << 2, phiIncr);
 
-	/* raise volume */
-	/* cs46xx_dsp_scb_set_volume (chip,src,0x7fff,0x7fff); */
-	
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 }
 
@@ -1641,9 +1635,8 @@
 								SCB_ON_PARENT_NEXT_SCB);
 
 
-	if (ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) 
-		/* set left (13), right validity bit (12) , and non-audio(1) and profsional bit (0) */
-		cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x00000000 | (1 << 13) | (1 << 12) | (1 << 1) | 1);
+	/* set spdif channel status value for streaming */
+	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_stream);
 
 	ins->spdif_status_out  |= DSP_SPDIF_STATUS_PLAYBACK_OPEN;
 
@@ -1659,7 +1652,7 @@
 	ins->spdif_status_out  &= ~DSP_SPDIF_STATUS_PLAYBACK_OPEN;
 
 	/* restore settings */
-	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x00000000 | (1 << 13) | (1 << 12));
+	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
 	
 	/* deallocate stuff */
 	cs46xx_dsp_remove_scb (chip,ins->asynch_tx_scb);

Re: Re: Phase Reversal on Santa Cruz (cs46xx)

[James Courtier-Dutton]

Just as a general point to note. We are not putting the audio card into AC3 mode.
We might be doing "passthru mode" or "spdif non-audio", but never just AC3 mode.
I help with the "xine" (a free media player http://xine.sf.net) that can play DVDs.
DVDs have AC3 audio tracks and DTS audio tracks.
xine uses "spdif non-audio" and "passthru mode" when used with alsa.
It can therefore output AC3 and DTS tracks to an external decoder.
Summary: -
AC3 is the wrong term to use. "spdif passthru" is probably a better term to use, as the current "ac3_mode" 
seems to imply that DTS audio might not work.
Cheers
James







-------------------------------------------------------
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Re: Re: Phase Reversal on Santa Cruz (cs46xx) => XFire too

[Friedrich Ewaldt]

  Hi Benny,

I noticed the phase reversal problem with my Terratec XFire, too: I 
played a mono wav. Then I combined the left and rigth output 
electrically and the sound was gone, i.e. one channel outputs the sound 
with a sign change, so that left+right=signal+(-signal)=0. The sound 
went back when unbalancing the output level. (This would also explain 
why my impulse response measuring program measures a negative response 
when connecting line out to line in ;-). I use the left channel of the 
soundcard for it.)
It applied your patch below, but it didn't solve the problem. Is it 
correct that the SDPIF AC3 mode switch is gone after applying the patch? 
I patched alsa version 0.9.0rc6.

fe

P.S. just tested my IR measuring program with right channel (and 
0.9.0rc6) => positive response
P.S.S. now I really wanted to know which channel has wrong sign with 
rc6. With 0.9.0rc2 I get positive impulse responses for both left and 
right channel.

conclusion: the left pcm channel of the alsa 0.9.0rc6 driver for cs46xx 
has wrong sign. Perhaps this can help a bit location the error.



Benny Sjostrand schrieb:

>>
>> no, I haven't, since the patch is quite mangled in the www archive 
>> and I wasn't subscribed then, I didn't go through th trouble of 
>> fixing it up. Is it merged into cvs now, and if not, could you send 
>> it my way? I'd be happy to give it a shot.
>>  
>>
>
> Here is the patch again, it's not in current CVS.
>
> /Benny
>
>------------------------------------------------------------------------
>
>diff --exclude=CVS -Naur alsa-kernel/include/cs46xx_dsp_spos.h ../cvs/alsa-kernel/include/cs46xx_dsp_spos.h
>--- alsa-kernel/include/cs46xx_dsp_spos.h	Sun Nov 10 19:47:01 2002
>+++ ../cvs/alsa-kernel/include/cs46xx_dsp_spos.h	Thu Nov 14 00:39:06 2002
>@@ -63,7 +63,6 @@
> #define DSP_SPDIF_STATUS_OUTPUT_ENABLED 1
> #define DSP_SPDIF_STATUS_PLAYBACK_OPEN  2
> #define DSP_SPDIF_STATUS_HW_ENABLED     4
>-#define DSP_SPDIF_STATUS_AC3_MODE       8
> 
> struct _dsp_module_desc_t;
> 
>@@ -196,6 +195,10 @@
> 	int spdif_status_in;
> 	u16 spdif_input_volume_right;
> 	u16 spdif_input_volume_left;
>+	/* spdif channel status,
>+	   left right and user validity bits */
>+	int spdif_csuv_default;
>+	int spdif_csuv_stream;
> 
> 	/* SPDIF input sample rate converter */
> 	dsp_scb_descriptor_t * spdif_in_src;
>diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/cs46xx_lib.c ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.c
>--- alsa-kernel/pci/cs46xx/cs46xx_lib.c	Mon Nov 11 23:04:50 2002
>+++ ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.c	Sun Nov 17 16:26:53 2002
>@@ -946,15 +946,15 @@
> 	case SNDRV_PCM_TRIGGER_START:
> 	case SNDRV_PCM_TRIGGER_RESUME:
> #ifdef CONFIG_SND_CS46XX_NEW_DSP
>+		/* magic value to unmute PCM stream  playback volume */
>+		snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 
>+				       SCBVolumeCtrl) << 2, 0x80007fff);
>+
> 		if (cpcm->pcm_channel->unlinked)
> 			cs46xx_dsp_pcm_link(chip,cpcm->pcm_channel);
>+
> 		if (substream->runtime->periods != CS46XX_FRAGS)
> 			snd_cs46xx_playback_transfer(substream, 0);
>-
>-		/* raise playback volume */
>-		cs46xx_dsp_scb_set_volume (chip,cpcm->pcm_channel->pcm_reader_scb,
>-					   chip->dsp_spos_instance->dac_volume_right,
>-					   chip->dsp_spos_instance->dac_volume_left);
> #else
> 		if (substream->runtime->periods != CS46XX_FRAGS)
> 			snd_cs46xx_playback_transfer(substream, 0);
>@@ -968,8 +968,9 @@
> 	case SNDRV_PCM_TRIGGER_STOP:
> 	case SNDRV_PCM_TRIGGER_SUSPEND:
> #ifdef CONFIG_SND_CS46XX_NEW_DSP
>-		/* mute channel */
>-		cs46xx_dsp_scb_set_volume (chip,cpcm->pcm_channel->pcm_reader_scb,0,0);
>+		/* magic mute channel */
>+		snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 
>+				       SCBVolumeCtrl) << 2, 0xffffffff);
> 
> 		if (!cpcm->pcm_channel->unlinked)
> 			cs46xx_dsp_pcm_unlink(chip,cpcm->pcm_channel);
>@@ -1063,6 +1064,7 @@
> }
> #endif
> 
>+
> static int snd_cs46xx_playback_hw_params(snd_pcm_substream_t * substream,
> 					 snd_pcm_hw_params_t * hw_params)
> {
>@@ -1079,15 +1081,10 @@
> 	down (&chip->spos_mutex);
> 
> 	snd_assert (cpcm->pcm_channel != NULL);
>-
>-	/* if IEC958 is opened in AC3 mode dont adjust SRCTask is not
>-	   used so dont adjust sample rate */
>-	if (cpcm->pcm_channel->pcm_channel_id != DSP_IEC958_CHANNEL ||
>-	    !(chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE)) {
>-		if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
>-			up (&chip->spos_mutex);
>-			return -ENXIO;
>-		}
>+	
>+	if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
>+		up (&chip->spos_mutex);
>+		return -ENXIO;
> 	}
> 
> 	if (cs46xx_dsp_pcm_channel_set_period (chip,cpcm->pcm_channel,period_size * 4)) {
>@@ -1886,8 +1883,8 @@
> {
> 	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
> 
>-	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_right;
>-	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_left;
>+	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_left;
>+	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_right;
> 
> 	return 0;
> }
>@@ -2038,34 +2035,6 @@
> 	return 0;
> }
> 
>-static int snd_cs46xx_iec958_ac3_mode_get(snd_kcontrol_t *kcontrol, 
>-					  snd_ctl_elem_value_t *ucontrol) 
>-{
>-	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
>-	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>-
>-	if (!ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) 
>-		ucontrol->value.integer.value[0] = 1;
>-	else
>-		ucontrol->value.integer.value[0] = 0;
>-
>-	return 0;
>-}
>-
>-static int snd_cs46xx_iec958_ac3_mode_put(snd_kcontrol_t *kcontrol, 
>-                                      snd_ctl_elem_value_t *ucontrol) 
>-{
>-	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
>-	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>-	int old = ins->spdif_status_out;
>-
>-	if (ucontrol->value.integer.value[0]) 
>-		ins->spdif_status_out |= DSP_SPDIF_STATUS_AC3_MODE;
>-	else
>-		ins->spdif_status_out &= ~DSP_SPDIF_STATUS_AC3_MODE;
>-
>-	return (old != ins->spdif_status_out);
>-}
> 
> static int snd_cs46xx_pcm_capture_put(snd_kcontrol_t *kcontrol, 
>                                       snd_ctl_elem_value_t *ucontrol)
>@@ -2127,6 +2096,110 @@
> 	return (val1 != snd_cs46xx_peekBA0(chip, BA0_EGPIODR));
> }
> 
>+
>+static int snd_cs46xx_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
>+{
>+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
>+	uinfo->count = 1;
>+	return 0;
>+}
>+
>+static int snd_cs46xx_spdif_default_get(snd_kcontrol_t * kcontrol,
>+					snd_ctl_elem_value_t * ucontrol)
>+{
>+	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
>+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>+
>+	down (&chip->spos_mutex);
>+	ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_default >> 24) & 0xff);
>+	ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_default >> 16) & 0xff);
>+	ucontrol->value.iec958.status[2] = 0;
>+	ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_default) & 0xff);
>+	up (&chip->spos_mutex);
>+
>+	return 0;
>+}
>+
>+static int snd_cs46xx_spdif_default_put(snd_kcontrol_t * kcontrol,
>+					snd_ctl_elem_value_t * ucontrol)
>+{
>+	cs46xx_t * chip = snd_kcontrol_chip(kcontrol);
>+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>+	unsigned int val;
>+	int change;
>+
>+	down (&chip->spos_mutex);
>+	val = _wrap_all_bits(((u32)ucontrol->value.iec958.status[0] << 24)) |
>+	      _wrap_all_bits(((u32)ucontrol->value.iec958.status[2] << 16)) |
>+	      _wrap_all_bits( (u32)ucontrol->value.iec958.status[3])  |
>+		/* left and right validity bit */
>+		(1 << 13) | (1 << 12);
>+
>+
>+	change = ins->spdif_csuv_default != val;
>+	ins->spdif_csuv_default = val;
>+
>+	if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) )
>+		cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val);
>+
>+	up (&chip->spos_mutex);
>+
>+	return change;
>+}
>+
>+static int snd_cs46xx_spdif_mask_get(snd_kcontrol_t * kcontrol,
>+				     snd_ctl_elem_value_t * ucontrol)
>+{
>+	ucontrol->value.iec958.status[0] = 0xff;
>+	ucontrol->value.iec958.status[1] = 0xff;
>+	ucontrol->value.iec958.status[2] = 0x00;
>+	ucontrol->value.iec958.status[3] = 0xff;
>+	return 0;
>+}
>+
>+static int snd_cs46xx_spdif_stream_get(snd_kcontrol_t * kcontrol,
>+                                         snd_ctl_elem_value_t * ucontrol)
>+{
>+	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
>+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>+
>+	down (&chip->spos_mutex);
>+	ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_stream >> 24) & 0xff);
>+	ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_stream >> 16) & 0xff);
>+	ucontrol->value.iec958.status[2] = 0;
>+	ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_stream) & 0xff);
>+	up (&chip->spos_mutex);
>+
>+	return 0;
>+}
>+
>+static int snd_cs46xx_spdif_stream_put(snd_kcontrol_t * kcontrol,
>+                                        snd_ctl_elem_value_t * ucontrol)
>+{
>+	cs46xx_t * chip = snd_kcontrol_chip(kcontrol);
>+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>+	unsigned int val;
>+	int change;
>+
>+	down (&chip->spos_mutex);
>+	val = _wrap_all_bits(((u32)ucontrol->value.iec958.status[0] << 24)) |
>+	      _wrap_all_bits(((u32)ucontrol->value.iec958.status[1] << 16)) |
>+	      _wrap_all_bits( (u32)ucontrol->value.iec958.status[3]) |
>+		/* left and right validity bit */
>+		(1 << 13) | (1 << 12);
>+
>+
>+	change = ins->spdif_csuv_stream != val;
>+	ins->spdif_csuv_stream = val;
>+
>+	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN )
>+		cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val);
>+
>+	up (&chip->spos_mutex);
>+
>+	return change;
>+}
>+
> #endif /* CONFIG_SND_CS46XX_NEW_DSP */
> 
> #ifdef CONFIG_SND_CS46XX_DEBUG_GPIO
>@@ -2238,7 +2311,7 @@
> },
> {
> 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
>-	.name = "IEC 958 Output Switch",
>+	.name = "IEC958 Output Switch",
> 	.info = snd_mixer_boolean_info,
> 	.get = snd_cs46xx_iec958_get,
> 	.put = snd_cs46xx_iec958_put,
>@@ -2246,14 +2319,7 @@
> },
> {
> 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
>-	.name = "IEC 958 AC3 Mode Switch",
>-	.info = snd_mixer_boolean_info,
>-	.get = snd_cs46xx_iec958_ac3_mode_get,
>-	.put = snd_cs46xx_iec958_ac3_mode_put,
>-},
>-{
>-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
>-	.name = "IEC 958 Input Switch",
>+	.name = "IEC958 Input Switch",
> 	.info = snd_mixer_boolean_info,
> 	.get = snd_cs46xx_iec958_get,
> 	.put = snd_cs46xx_iec958_put,
>@@ -2261,12 +2327,34 @@
> },
> {
> 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
>-	.name = "IEC 958 Input Volume",
>+	.name = "IEC958 Input Volume",
> 	.info = snd_cs46xx_vol_info,
> 	.get = snd_cs46xx_vol_iec958_get,
> 	.put = snd_cs46xx_vol_iec958_put,
> 	.private_value = (ASYNCRX_SCB_ADDR + 0xE) << 2,
> },
>+{
>+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
>+	.name =  SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
>+	.info =	 snd_cs46xx_spdif_info,
>+	.get =	 snd_cs46xx_spdif_default_get,
>+	.put =   snd_cs46xx_spdif_default_put,
>+},
>+{
>+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
>+	.name =	 SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
>+	.info =	 snd_cs46xx_spdif_info,
>+        .get =	 snd_cs46xx_spdif_mask_get,
>+	.access = SNDRV_CTL_ELEM_ACCESS_READ
>+},
>+{
>+	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
>+	.name =	 SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
>+	.info =	 snd_cs46xx_spdif_info,
>+	.get =	 snd_cs46xx_spdif_stream_get,
>+	.put =	 snd_cs46xx_spdif_stream_put
>+},
>+
> #endif
> #ifdef CONFIG_SND_CS46XX_DEBUG_GPIO
> {
>diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/cs46xx_lib.h ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.h
>--- alsa-kernel/pci/cs46xx/cs46xx_lib.h	Sun Nov 10 19:47:01 2002
>+++ ../cvs/alsa-kernel/pci/cs46xx/cs46xx_lib.h	Wed Nov 13 22:51:21 2002
>@@ -217,6 +217,6 @@
> 							       int period_size);
> int                        cs46xx_dsp_pcm_ostream_set_period (cs46xx_t * chip,
> 							      int period_size);
>-int                        cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 right,u16 left);
>-int                        cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 right,u16 left);
>+int                        cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 left,u16 right);
>+int                        cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 left,u16 right);
> #endif /* __CS46XX_LIB_H__ */
>diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos.c ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.c
>--- alsa-kernel/pci/cs46xx/dsp_spos.c	Sun Nov 10 19:47:01 2002
>+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.c	Thu Nov 14 19:59:59 2002
>@@ -31,6 +31,7 @@
> #include <sound/core.h>
> #include <sound/control.h>
> #include <sound/info.h>
>+#include <sound/asoundef.h>
> #include <sound/cs46xx.h>
> 
> #include "cs46xx_lib.h"
>@@ -262,6 +263,15 @@
> 	ins->spdif_input_volume_right = 0x8000;
> 	ins->spdif_input_volume_left = 0x8000;
> 
>+	/* set left and right validity bits and
>+	   default channel status */
>+	ins->spdif_csuv_default = 
>+		ins->spdif_csuv_stream =  
>+	 /* byte 0 */  (_wrap_all_bits(  (SNDRV_PCM_DEFAULT_CON_SPDIF        & 0xff)) << 24) |
>+	 /* byte 1 */  (_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 16) & 0xff)) << 16) |
>+	 /* byte 3 */   _wrap_all_bits(  (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) |
>+	 /* left and right validity bits */ (1 << 13) | (1 << 12);
>+
> 	return ins;
> }
> 
>@@ -1549,7 +1559,7 @@
> 	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CONTROL, 0x80000000);
> 
> 	/* right and left validate bit */
>-	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x00000000 | (1 << 13) | (1 << 12));
>+	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
> 
> 	/* monitor state */
> 	ins->spdif_status_out |= DSP_SPDIF_STATUS_HW_ENABLED;
>@@ -1587,11 +1597,6 @@
> 	cs46xx_poke_via_dsp (chip,SP_SPDIN_FIFOPTR, 0x0);
> 	cs46xx_src_link(chip,ins->spdif_in_src);
> 
>-	/* restore SPDIF input volume */
>-	cs46xx_dsp_scb_set_volume (chip,ins->spdif_in_src,
>-				   ins->spdif_input_volume_right,
>-				   ins->spdif_input_volume_left);
>-
> 	spin_unlock_irq(&chip->reg_lock);
> 
> 	/* set SPDIF input sample rate and unmute
>@@ -1725,39 +1730,47 @@
> 	return 0;
> }
> 
>-int cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 right,u16 left)
>+int cs46xx_dsp_set_dac_volume (cs46xx_t * chip,u16 left,u16 right)
> {
>-	int i;
> 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
>+	dsp_scb_descriptor_t * scb; 
> 
> 	down(&chip->spos_mutex);
>+	
>+	/* main output */
>+	scb = ins->master_mix_scb->sub_list_ptr;
>+	while (scb != ins->the_null_scb) {
>+		cs46xx_dsp_scb_set_volume (chip,scb,left,right);
>+		scb = scb->next_scb_ptr;
>+	}
> 
>-	ins->dac_volume_right = right;
>-	ins->dac_volume_left = left;
>-
>-	for (i = 0; i < DSP_MAX_PCM_CHANNELS; ++i) {
>-		if (ins->pcm_channels[i].active &&
>-		    !ins->pcm_channels[i].unlinked) {
>-			cs46xx_dsp_scb_set_volume (chip,ins->pcm_channels[i].pcm_reader_scb,
>-						   right,left);
>-			
>-		}
>+	/* rear output */
>+	scb = ins->rear_mix_scb->sub_list_ptr;
>+	while (scb != ins->the_null_scb) {
>+		cs46xx_dsp_scb_set_volume (chip,scb,left,right);
>+		scb = scb->next_scb_ptr;
> 	}
> 
>+	ins->dac_volume_left = left;
>+	ins->dac_volume_right = right;
>+
> 	up(&chip->spos_mutex);
> 
> 	return 0;
> }
> 
>-int cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 right,u16 left) {
>+int cs46xx_dsp_set_iec958_volume (cs46xx_t * chip,u16 left,u16 right) {
> 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
> 
> 	down(&chip->spos_mutex);
>-	cs46xx_dsp_scb_set_volume (chip,ins->spdif_in_src,
>-				   right,left);
> 
>-	ins->spdif_input_volume_right = right;
>+	if (ins->asynch_rx_scb != NULL)
>+		cs46xx_dsp_scb_set_volume (chip,ins->asynch_rx_scb,
>+					   left,right);
>+
> 	ins->spdif_input_volume_left = left;
>+	ins->spdif_input_volume_right = right;
>+
> 	up(&chip->spos_mutex);
> 
> 	return 0;
>diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos.h ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.h
>--- alsa-kernel/pci/cs46xx/dsp_spos.h	Sun Nov 10 19:47:01 2002
>+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos.h	Thu Nov 14 09:36:26 2002
>@@ -185,6 +185,25 @@
> #define SP_SPDOUT_CONTROL 0x804D
> #define SP_SPDOUT_CSUV    0x808E
> 
>+static inline u8 _wrap_all_bits (u8 val) {
>+	u8 wrapped;
>+	
>+	/* wrap all 8 bits */
>+	wrapped = 
>+		((val & 0x1 ) << 7) |
>+		((val & 0x2 ) << 5) |
>+		((val & 0x4 ) << 3) |
>+		((val & 0x8 ) << 1) |
>+		((val & 0x10) >> 1) |
>+		((val & 0x20) >> 3) |
>+		((val & 0x40) >> 5) |
>+		((val & 0x80) >> 6);
>+
>+	return wrapped;
>+
>+}
>+
>+
> static inline void cs46xx_dsp_spos_update_scb (cs46xx_t * chip,dsp_scb_descriptor_t * scb) 
> {
> 	/* update nextSCB and subListPtr in SCB */
>@@ -195,12 +214,11 @@
> }
> 
> static inline void cs46xx_dsp_scb_set_volume (cs46xx_t * chip,dsp_scb_descriptor_t * scb,
>-					      u16 right,u16 left) {
>-	unsigned int val = ((0xffff - right) << 16 | (0xffff - left));	
>+					      u16 left,u16 right) {
>+	unsigned int val = ((0xffff - left) << 16 | (0xffff - right));
> 
> 	snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val);
> 	snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val);
> }
>-
> #endif /* __DSP_SPOS_H__ */
> #endif /* CONFIG_SND_CS46XX_NEW_DSP  */
>diff --exclude=CVS -Naur alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c ../cvs/alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c
>--- alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c	Sun Nov 10 19:47:01 2002
>+++ ../cvs/alsa-kernel/pci/cs46xx/dsp_spos_scb_lib.c	Thu Nov 14 09:08:14 2002
>@@ -603,8 +603,8 @@
> 		src_buffer_addr << 0x10,
> 		0x04000000,
> 		{ 
>-			0x8000,0x8000,
>-			0xffff,0xffff
>+			0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left,
>+			0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left
> 		}
> 	};
> 
>@@ -658,7 +658,7 @@
> 		/* D */ 0,
> 		{
> 			/* E */ 0x8000,0x8000,
>-			/* F */ 0xffff,0xffff
>+			/* F */ 0x8000,0x8000
> 		}
> 	};
> 
>@@ -830,7 +830,7 @@
> 		0,0x2aab,           /* Const 1/3 */
>     
> 		{
>-			0,                /* Define the unused elements */
>+			0,         /* Define the unused elements */
> 			0,
> 			0
> 		},
>@@ -846,7 +846,7 @@
> 		   rate etc  */
> 		0x18000000,                     /* Phi increment for approx 32k operation */
> 		0x8000,0x8000,                  /* Volume controls are unused at this time */
>-		0xffff,0xffff
>+		0x8000,0x8000
> 	};
>   
> 	scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_tx_scb,
>@@ -864,7 +864,7 @@
>                                    dsp_scb_descriptor_t * parent_scb,
>                                    int scb_child_type)
> {
>-
>+	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
> 	dsp_scb_descriptor_t * scb;
> 
> 	asynch_fg_rx_scb_t asynch_fg_rx_scb = {
>@@ -893,9 +893,9 @@
> 		   rate etc  */
> 		0x18000000,         
> 
>-		/* Mute stream */
>-		0x8000,0x8000,       
>-		0xffff,0xffff
>+		/* Set IEC958 input volume */
>+		0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left,
>+		0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left,
> 	};
> 
> 	scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_rx_scb,
>@@ -1116,11 +1116,13 @@
> 	case DSP_IEC958_CHANNEL:
> 		snd_assert (ins->asynch_tx_scb != NULL, return NULL);
> 		mixer_scb = ins->asynch_tx_scb;
>+#if 0
> 		if (ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) {
> 			snd_printdd ("IEC958 opened in AC3 mode\n");
> 			/*src_scb = ins->asynch_tx_scb;
> 			  ins->asynch_tx_scb->ref_count ++;*/
> 		}
>+#endif
> 		break;
> 	default:
> 		snd_assert (0);
>@@ -1198,9 +1200,7 @@
> 			return NULL;
> 		}
> 
>-		if (pcm_channel_id != DSP_IEC958_CHANNEL ||
>-		    !(ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE))
>-			cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate);
>+		cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate);
> 
> 		ins->nsrc_scb ++;
> 	} 
>@@ -1461,17 +1461,11 @@
> 	 */
> 	spin_lock_irqsave(&chip->reg_lock, flags);
> 
>-	/* mute SCB */
>-	/* cs46xx_dsp_scb_set_volume (chip,src,0,0); */
>-
> 	snd_cs46xx_poke(chip, (src->address + SRCCorPerGof) << 2,
> 	  ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));
> 
> 	snd_cs46xx_poke(chip, (src->address + SRCPhiIncr6Int26Frac) << 2, phiIncr);
> 
>-	/* raise volume */
>-	/* cs46xx_dsp_scb_set_volume (chip,src,0x7fff,0x7fff); */
>-	
> 	spin_unlock_irqrestore(&chip->reg_lock, flags);
> }
> 
>@@ -1641,9 +1635,8 @@
> 								SCB_ON_PARENT_NEXT_SCB);
> 
> 
>-	if (ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) 
>-		/* set left (13), right validity bit (12) , and non-audio(1) and profsional bit (0) */
>-		cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x00000000 | (1 << 13) | (1 << 12) | (1 << 1) | 1);
>+	/* set spdif channel status value for streaming */
>+	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_stream);
> 
> 	ins->spdif_status_out  |= DSP_SPDIF_STATUS_PLAYBACK_OPEN;
> 
>@@ -1659,7 +1652,7 @@
> 	ins->spdif_status_out  &= ~DSP_SPDIF_STATUS_PLAYBACK_OPEN;
> 
> 	/* restore settings */
>-	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x00000000 | (1 << 13) | (1 << 12));
>+	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
> 	
> 	/* deallocate stuff */
> 	cs46xx_dsp_remove_scb (chip,ins->asynch_tx_scb);
>  
>





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Re: Re: Phase Reversal on Santa Cruz (cs46xx) => XFire too

[Kevin Puetz]

On Tuesday 03 December 2002 16:24, Friedrich Ewaldt wrote:
>   Hi Benny,

w00t!

it seems that volumes 0-0x7fff ramp up volumes, then 8000-ffff go back down... 
but with opposite phases. So...

--- alsa-cvs/alsa-driver-0.9.0rc6cvs20021202/alsa-kernel/pci/cs46xx/dsp_spos.h	
2002-11-24 19:52:17.000000000 -0600
+++ modules/alsa-driver/alsa-kernel/pci/cs46xx/dsp_spos.h	2002-12-03 
17:27:10.000000000 -0600
@@ -215,7 +215,7 @@
 
 static inline void cs46xx_dsp_scb_set_volume (cs46xx_t * 
chip,dsp_scb_descriptor_t * scb,
 					      u16 left,u16 right) {
-	unsigned int val = ((0xffff - left) << 16 | (0xffff - right));
+	unsigned int val = (left << 16 | (0xffff - right));
 
 	snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val);
 	snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val);


and it sounds correct (for CD analog passthru and PCM at least).
(note - you'll need to open alsamixer and adjust the DAC volume before the 
'fix' works, I haven't found where it gets programmed on driver load.

Now, I have no idea if this is correct, if I just broke SPDIF (no reciever), 
rear speakers (ditto, I don't have any) but it fixes my case anyway, so 
that's a start. If anyone can test more of these cases, that might be good.

Throwing it out before I head out here for a few hours, in the hope that it 
will cause some insight :-)


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Re: Re: Phase Reversal on Santa Cruz (cs46xx) => XFire too

[Benny Sjostrand]

>
>
>it seems that volumes 0-0x7fff ramp up volumes, then 8000-ffff go back down... 
>but with opposite phases. So...
>
>  
>
Great discovery!, probably the stream can be inverted several times when 
it processedthrough
through the SCB tree. (PCMReader->SrcTaskSCB->MasterMixSCB->CodecOut)
Almost all SCB's got a volume field.

>and it sounds correct (for CD analog passthru and PCM at least).
>(note - you'll need to open alsamixer and adjust the DAC volume before the 
>'fix' works, I haven't found where it gets programmed on driver load.
>  
>
I guess that the only thing left is to change the initial volume values 
set when driver is loaded
in dsp_spos.c line 260:
    /* maximize volume */
     ins->dac_volume_right = 0x8000;
-    ins->dac_volume_left = 0x8000;
+   ins->dac_volume_left = 0x7fff;
     ins->spdif_input_volume_right = 0x8000;
-    ins->spdif_input_volume_left = 0x8000;
+   ins->spdif_input_volume_left = 0x7fff;

>Now, I have no idea if this is correct, if I just broke SPDIF (no reciever), 
>rear speakers (ditto, I don't have any) but it fixes my case anyway, so 
>that's a start. If anyone can test more of these cases, that might be good.
>
>  
>
If it works and solves a problem then it must correct -:)


/Benny



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Re: Re: Phase Reversal on Santa Cruz (cs46xx) => XFire too

[Friedrich Ewaldt]

  Hi Kevin, hi Benny,

with your changes applied, now both PCM channels have the same phase 
with my XFire. The stereo image makes much more sense that way :-). 
Thanks a lot!

What I've noticed: Switching the 'PCM Out' control from 'pre 3D' to 
'post 3D' inverts the sign of both l&r channels. (behaviour is 
independent of '3D depth', '3D center' and '3D switch' controls). But 
that doesn't cause any problems, at least with stereo output (which is 
the only mode the XFire works for now). With >2 channel output on other 
cs46xx cards this could perhaps cause problems if the phase of the 
rear/LFE/center channels doesn't match the left/right channel phase any 
more. But I can't check this with my card.

@Benny:
Another problem: SPDIF input doesn't work anymore since I applied the 
patch attached in your mail on 11/23/02. There is no 'SPDIF ac3 mode' in 
alsamixer after patching. Is this correct? And I noticed that I can't 
change the SPDIF input volume for left and right channel independently 
anymore. If I move the slider for one channel, the other follows more or 
less. I get a maximum volume difference of 4 (on the range 0..100) 
between left and right SPDIF input channel. Changing the levels 
independently and SPDIF input worked before patching.
I reinstalled the unpatched rc6 driver but even restarting alsa didn't 
make SPDIF input working again (I also deleted my /etc/asound.state 
because the mixer controls have changed). Finally, I found out that I 
had to switch off my PC for several seconds and restart with the 
unpatched rc6 driver. The patched driver changes the state of the 
soundcard in some way that even reinstalling the unpatched one doesn't 
help without cold booting. Does SPDIF input work for other cs46xx based 
cards after applying the mentioned patch? What errors does this patch 
fix? Do I need it?

Anyway, great to have pcm output without phase reversals. Thanks again!
fe


Benny Sjostrand schrieb:

>>
>>
>> it seems that volumes 0-0x7fff ramp up volumes, then 8000-ffff go 
>> back down... but with opposite phases. So...
>>
>>  
>>
> Great discovery!, probably the stream can be inverted several times 
> when it processedthrough
> through the SCB tree. (PCMReader->SrcTaskSCB->MasterMixSCB->CodecOut)
> Almost all SCB's got a volume field.
>
>> and it sounds correct (for CD analog passthru and PCM at least).
>> (note - you'll need to open alsamixer and adjust the DAC volume 
>> before the 'fix' works, I haven't found where it gets programmed on 
>> driver load.
>>  
>>
> I guess that the only thing left is to change the initial volume 
> values set when driver is loaded
> in dsp_spos.c line 260:
>    /* maximize volume */
>     ins->dac_volume_right = 0x8000;
> -    ins->dac_volume_left = 0x8000;
> +   ins->dac_volume_left = 0x7fff;
>     ins->spdif_input_volume_right = 0x8000;
> -    ins->spdif_input_volume_left = 0x8000;
> +   ins->spdif_input_volume_left = 0x7fff;
>
>> Now, I have no idea if this is correct, if I just broke SPDIF (no 
>> reciever), rear speakers (ditto, I don't have any) but it fixes my 
>> case anyway, so that's a start. If anyone can test more of these 
>> cases, that might be good.
>>
>>  
>>
> If it works and solves a problem then it must correct -:)
>
>
> /Benny






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